1. Technical Field
The present invention is related to a network system integrated with SIP call server and SIP agent client, and more particularly to a network system which is integrated with SIP call server and SIP agent client based on Session Initiation Protocol (SIP).
2. Description of the Prior Art
Many new techniques have been developed along with the popularity of advanced Internet. For example, voice over Internet protocol (VoIP) is proposed to provide voice telecommunication on IP network of LAN or Internet as using phone. This can save a great deal of calling fee. Such technique includes PC-to-PC telecommunication between two surfing computers, PC-to-phone telecommunication between a surfing computer and an ordinary phone via an integrated networked private branch exchange (PBX), phone-to-phone telecommunication between two phones via VoIP gateway of network telephone company for network digitalized conversion and device-to-device telecommunication between two IP phones.
The basic operation principle of VoIP is to transmit analog voice signal from a telephone, facsimile or PBX to a router for converting and compressing the voice signal into a data packet. The data packet is transmitted to a remote router via IP network. The remote router converts the data packet back into analog voice signal and then transmits the signal to the telephone, facsimile or PBX. Finally, the signal is sent to a user end. Accordingly, via the Internet, the remote telecommunication can be performed all over the world without using the conventional public telephone network (PSTN).
However, the existent VoIP technique such H323/H248, etc. regulated by International Telecommunication Union (ITU) is designed for local area network and is not fully applicable to the open environment of Internet. Moreover, the VoIP technique involves complicated structure and more strict limits. Therefore, the conversion between the existent VoIP technique and the PSTN is relatively complicated. In order to solve the above problems, Internet Engineering Task Force (IETF) has developed a new protocol, that is, Session Initiation Protocol (SIP). This protocol is fully applicable to the integrated environment of Internet and PSTN.
The SIP pertains to an application layer protocol in the seven-layer structure of open system interface (OSI) as the client-server structure of HTTP protocol. In packet processing, the commands and states can be transmitted in pure text by means of the read packet data of HTTP. Therefore, the SIP is very suitable for the transmission architecture of wide area network.
In the SIP structure, at least one SIP call server must be built in addition to the user agent (UA). The SIP call server can serve as a proxy server, redirect server, registry server, voice mail server, etc. The SIP call server is functionally an integrated software and can be combined with the existent PSTN, VoIP, etc.
However, in the SIP architecture, each UA must register one's own SIP URI and current IP location in the registry server, whereby the SIP call server can identify every UA. After registered, other UA on the Internet can communicate with the UA through the SIP call server.
In addition, the SIP pertains to application layer protocol so that the software can be easily developed independent to lower layer transmission or network. Therefore, the SIP can be built on various networks or servers. Also, it is very easy to integrate the respective systems. For example, the SIP can be integrated with the internal server, database, WWW website, chat room or video meeting system of a corporation. Alternatively, the SIP can be integrated with external PSTN or VoIP.
It can be known from the above that the SIP is advantageous with easy integration and reduced telecommunication fee. Therefore, a company can utilize the internal broad band and external broad band network between the headquarter and branches to build the SIP telecommunication network. Accordingly, the remote telecommunication fee or international telecommunication fee between the headquarter and the remote branches or oversea branches can be reduced.
The basic architecture of the SIP includes at least one SIP call server, at least one local user agent client of VoIP and at least one remote user agent client of IP phone. Therefore, presently, when building the SIP telecommunication network, it is still necessary to register on the external SIP call server. This is inconvenient. Moreover, in the case that there are a number of registered users, the user still needs to afford the fees for the registry and bridge telecommunication.
Therefore, the applicant is intended to develop a network apparatus integrated with SIP call server and SIP agent client. The SIP call server and the SIP agent client are integrated with a broad band network apparatus such as ADSL modem, network hub, wireless LAN access point, IP sharer, etc. By means of the present invention, a user by oneself can build a private SIP telecommunication network to eliminate the troublesome registration on external SIP call server. In addition, the cost for building the SIP call server is saved and the telecommunication fee is greatly reduced.